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[Wednesday at NAB]
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AoIP Changes Audio Delivery, On-Air Calls
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by Timothy Kimble,
~ April 16, 2008
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RADIO WORLD
Codecs, microwave hops, frame relay systems, T1 lines, ISDN — all were “the new thing” at one time or another, all are old hat by now.
The new, inexpensive, easy-to-use kid on the block for getting “here to there” is audio over IP. It’s now been around for a while too; but acceptance continues to spread and broadcast manufacturers continue to find cheaper, more reliable ways to use IP to transport audio from studio to studio, remote to control room and even studio to transmitter.
Even though it may seem like an impossible amount of new ideas are born every year, some industry leaders do believe research and development won’t keep its hectic pace.
“I do think the pace of development will eventually slow down, as the market matures, and customers will probably standardize on a couple of variants,” said Barix CEO Johannes Reitschel.
Still, Barix has updates for its Exstreamer line, and is showing the Exstreamer-110 IP audio decoder at the NAB Show. Its previous Exstreamer-100 offered transport of MPEG-4. The newer 110 offers AAC audio, a much higher standard with digital clarity.
“I guess I don’t have to tell you about the advantages over the MP3 codec: much better audio quality at similar bit rates, or similar audio quality at much lower bit rates,” sasid Reitschel. Barix also is debuting software that adds store-and-forward message repeating for oft-repeated bits of audio like legal identification announcements.
Telos Systems is at the show with several offerings in the telco and codec product class.
Its Nx12 talkshow system supports ISDN, the latest hybrid technology and Livewire audio-over-IP. It is a self-contained 12-line system that includes four hybrids, and it is offered with Livewire and either analog or AES inputs/outputs. Each hybrid has its own Omnia AGC and noise gate. The system includes echo cancellation to help with VoIP and cellular callers.
Telos is also showing the new Zephyr/IP audio codec, optimized for operation over the public Internet and mobile phone data services. It uses Agile Connection Technology, which combines loss detection and concealment with dynamic buffering and auto-varying bitrate functions. Telos says this means Z/IP continuously adapts to network conditions, minimizing effects of packet loss, varying bandwidth and jitter. Z/IP also works with high-speed mobile phone data networks; a PC Card slot accepts standard mobile data cards.
Z/IP uses AAC-ELD, a new codec based on low-delay AAC. Telos says it delivers superior audio for two-way IP applications over non-controlled networks.
A CALLER IN EVERY POTS
Something old is new again with a telephone hybrid system from AVT. It has come up with a new version of its Magic line, which offers POTS, plain old telephone service.
The Magic POTS system handles up to 16 callers, and can be operated via LAN. It comes with screening software that includes a database and is optimized for touchscreen operations. The new technology will make the lives of producers putting through multiple calls much easier.
In conference mode, callers are digitally mixed into the system and mix-minus signals are generated for the callers. “The software is adjustable to specific situations,” said AVT marketing guru Ulrike Lauterbach. “To wrap it up, the system is flexible, user-friendly and a high-quality product.” A smaller version, the Magic TH2 POTS system also will be on display.
AVT also is showing new IP codecs. The Magic AC1 XIP and Magic AC1 XIP RM offer IP, ISDN and X.21 service, and can be installed in the studio because they operate quietly without a fan. The units provide g7.11 and g7.22, and linear PCM. The RM is a standard rack-mount unit, while the XIP is a compact unit, offering the same features. Alongside that will be the Magic AE1 DAB+ audio encoder, which offers the higher-quality AACv2 format.
Comrex is at the NAB Show with the DH42, a digital telco hybrid that accommodates two traditional POTS lines as well as two voice-over-IP lines in a single RU enclosure.
“Essentially a four-line conferencing hybrid, the DH42 allows callers to be put on-air with separate send and receive paths, filtering, AGC and control functions,” the company said. The unit offers broadcasters the ability to interface their Internet-based VoIP lines or VoIP PBX lines with audio consoles and other broadcast audio gear.
Tieline Technology is demonstrating its QOS Performance Engine Technology over IP for the G3 line of audio codecs. The company says this provides greater connection stability with less delay for live remote broadcasts over IP networks.
Noting that the Net and 3G wireless networks are lossy packet-switched networks in which a percentage of data packets sent never arrive — 1 to 3 three percent of packets over open Internet IP connections and up to 10 percent in 3G networks — Tieline said its QOS provides reliable managed audio over these networks.
“Our new IP QOS Performance Engine automatically manages the IP connection,” said Darren Levy. “It dynamically adjusts settings in the codec to allow for varying packet arrival times as well as providing multi-strategy Forward Error Correction (FEC) analysis, which replaces lost or corrupted packets in a data stream. Tieline G3 codecs are able to react intuitively to variations in packet-switched network conditions and virtually guarantee a solid connection.”
AEQ is displaying its Phoenix IP audio codec family. The Phoenix Mobile is portable, as the name suggests, while the Phoenix Studio is for back home in the rack, a 1U configuration. The Phoenix codecs have two expansion slots of optional modules, the first of which will be a POTS module. ISDN, X2.1, V35 and GSM will be available.
Linear Acoustic is showcasing its Stream Stacker-HD e2, an update of its high-density audio distribution system that adds the ability to distribute Dolby-compatible metadata on multiple programs. The company says improvements also have been made in jitter performance and increased error resilience.
Mayah Communications is featuring its Flashman II, a portable recorder and codec. It’s a bit larger than your typical Walkman-type recorder thanks to the two phantom-powered XLR microphone inputs. It records WAV, MPEG Layer II and Layer III and AACv2. The Flashman II can connect via Ethernet or any type of IP network. Optional cards support LAN connectivity.
Mayah’s Sporty reporter codec can do much of the same, with flexible presets for an on-board mixer, represented by a backlit display on the front of the unit. On top is a phone pad, and it offers the same audio formats as the Flashman II, with additions helpful to those reporting from the field. Get access to an Internet connection and you can send digital-quality audio back to the station. Sporty also has ISDN capability.
APT is launching an audio codec designed for HD Radio. The World Cast Horizon HD has T1 and IP ports to deliver content to both the FM and HD1 streams. It has an RS-232 port to enter PAD content or contact closures for remote control.
Portability also is a key feature on a new Musicam USA product, the Road Warrior. It’s a compact version of the company’s Suprima audio codec with a mixer added on. Both offer audio standard from MPEG Layer II up through AAC, and connectivity over IP or ISDN.
Additionally, the SupriMax rack-mount can hold 14 of the Suprima codec modules in a 3U space. For smaller operations, the SupriMax 1U holds four of the modules in, you guessed it, 1U.
These new devices represent a continuing massive change in how audio is delivered and how callers get on the air. Many of them have taken much of the guess work producers and engineers had to contend with way back in the late 20th century.
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